Sound signal processing apparatus and sound signal processing method

ABSTRACT

A sound signal processing apparatus includes a sound source direction determination unit and a filter processing unit. The sound source direction determination unit determines sound source directions with respect to sound signals of a plurality of channels for respective first to n-th bands. The filter processing unit includes first to n-th filters which are connected in series and configured to boost or attenuate the sound signals with respect to the first to n-th bands. The respective first to n-th filters perform boosting or attenuation based on the sound source directions of the first to n-th bands which are determined by the sound source direction determination unit.

BACKGROUND

The present disclosure relates to a sound signal processing apparatusand a sound signal processing method for obtaining a sound from aspecific sound source direction.

Japanese Unexamined Patent Application Publication No. 2010-11117 andJapanese Unexamined Patent Application Publication No. 2007-129383 areexamples of related art.

For example, a beam forming technique for forming directivity withrespect to input sounds from two microphones is known.

FIG. 10 illustrates an example of a noise cancellation headphone(hereinafter, referred to as an NC headphone). Although an NC headphone100 supplies stereo sounds to a user using left and right speakers 101Land 101R, microphones 102L and 102R that absorb external sounds areprovided in order to reduce external noises.

The NC headphone 100, for example, reproduces and outputs the sounds ofreproduction music from a portable media player or the like.

In brief, in order to cancel noises, reversed phase components of soundsignals absorbed by the microphones 102L and 102R are generated,combined with respective music signals, and then output from thespeakers 101L and 101R. Therefore, the sounds of the music signals arelistened to by a user in a state in which external noises are spatiallycancelled.

Here, it is considered that the microphones 102L and 102R are used notonly for canceling noise but also for absorbing external sounds whichhave directivity.

For example, although it is preferable that a user can normally performconversation or the like even when wearing the NC headphone 100, if anoise cancellation function is turned on, for example, even the soundsof a person who is in front of the user are reduced, so that it isdifficult to listen to conversational sounds.

Therefore, for example, when a conversation or the like is performed, amode that turns off reproduction music and turns off a noisecancellation function is provided.

However, if the noise cancellation function is turned off, surroundingnoise is heard to a large degree together with the sounds of otherpeople. Therefore, in a place where there is much traffic, the inside ofa plane, or the like, a state in which conversational sounds or the likeare difficult to be heard is not changed.

In such a case, it is preferable that a speaker output, in whichconversational sounds are easily heard and surrounding noises aresuppressed, can be realized.

If it is considered that a user wears the NC headphone 100 and faces thefront as in FIG. 10, it can be considered that the sounds of a targetthat conducts conversation come from the front of the user in mostcases. At this time, as shown in FIG. 10, when viewed from the user, theuser regards sound sources other than from the front as noises, of whichthe level thereof should be lowered while boosting the conversationalsounds from the front.

In order to realize this, when a necessary sound source direction istemporarily set to the front, directivity can be formed at the time ofabsorbing sounds using a so-called beam forming method.

FIG. 11A is a conceptual diagram illustrating a beam forming process,and sound signals from left and right microphones 102L and 102R areprocessed and output by a beam forming process unit 103.

The simplest beam forming process may be a process of adding soundsignals from the left and right microphones 102L and 102R as shown inFIG. 11B when the necessary directivity is the front or the back.

In this case, the phases of the sound signal components of left andright channels with respect to sounds from the front or back, that is,sounds from the sound sources at an equal distance from the microphones102L and 102R, are matched with each other, and boosted by addition.Since the phases of the sound signal components of sounds from otherdirections are deviated from the phases of the sound signal componentsof the left and right channels, the sound signal components are reducedby as much as the deviation. Therefore, sound signals having, forexample, directivity in the front direction can be obtained.

Meanwhile, the beam forming process itself can boost the sound signalsin the directions other than the front direction. In this case, a delayunit is installed on one side channel, with the result that the timedifference between the same wave fronts which reach each of themicrophones can be absorbed, so that the beam forming can be realized inan oblique direction or in a traverse direction.

In order to increase the accuracy of the beam forming (in this case, thesame meaning as the boosting of the front directivity and the reductionin surrounding noises), a noise suppression device that mainly uses bandpass filters shown in FIG. 12 is generally used and not the simpledevice as shown in FIG. 11B.

The sound signal obtained by the microphone 102L is amplified by themicrophone amplifier 104L, and then supplied to band pass filters 121L,122L, and 123L that have central pass frequencies fc1, fc2, and fc3,respectively. In the band pass filters 121L, 122L, and 123L, the soundsignal components of the bands BD1, BD2, and BD3 are extracted.

Further, the sound signal obtained by the microphone 102R is amplifiedby the microphone amplifier 104R, and then supplied to band pass filters121R, 122R, and 123R that have central pass frequencies fc1, fc2, andfc3, respectively, so that the sound signal components of the respectivebands BD1, BD2, and BD3 are extracted.

Meanwhile, the pass band of the band pass filter that has the centralfrequency fc1 is represented as a band BD1. In the same manner, the passbands of the band pass filters that have the central frequencies fc2 andfc3 are represented as bands BD2 and BD3.

The sound signal components of the band BD1, which are the outputs ofthe band pass filters 121L and 121R, are supplied to the sound sourcedirectional angle analysis unit 124 and the adder 127.

The sound signal components of the band BD2, which are the output of theband pass filters 122L and 122R, are supplied to the sound sourcedirectional angle analysis unit 125 and the adder 128.

The sound signal components of the band BD3, which are the outputs ofthe band pass filters 123L and 123R, are supplied to the sound sourcedirectional angle analysis unit 126 and the adder 129.

The sound source directional angle analysis units 124, 125, and 126determine the sound source direction of a dominant sound from among thesound signal components of the bands BD1, BD2, and BD3, respectively.

Thereafter, the sound source directional angle analysis units 124, 125,and 126 control the gain of the variable gain amplifiers 130, 131, and132 based on the determined direction. That is, the sound sourcedirectional angle analysis units 124, 125, and 126 perform control suchthat the gain increases when the determined direction is a targetdirection, such as the front direction or the like, and such that thegain decrease when the determined direction is the other direction.

Each of the sound signal components of the bands BD1, BD2, and BD3 isadded by the adders 127, 123, and 129 for the respective L and Rchannels, and then supplied to the variable gain amplifiers 130, 131,and 132. Thereafter, the variable gain amplifiers 130, 131, and 132 arecontrolled by the sound source directional angle analysis units 124,125, and 126 as described above, so that, for example, a band in whichthe sound from the front direction is dominant is boosted, and the otherbands are reduced. The outputs of the respective bands BD1, BD2, and BD3in which gains are adjusted as weights for respective bands are added byan adder 133, and become an output sound signal Sout on which a beamforming process has been performed.

When a beam forming process unit 103 using such a noise suppressiondevice is used, the conversation sounds are not easily buried in noisesand can be heard in the state as shown in FIG. 10.

Further, as one type of method of boosting sounds and suppressingnoises, a method using an FFT that centers on “spectrum subtraction” maybe provided as a representative method of analyzing and combining soundswithout using the beam forming method in order to remove noises in therelated art.

SUMMARY

As described above, as the representative method of analyzing andcombining sounds in order to reduce noises in the related art, twomethods that use band pass filters and FFTs respectively are provided.

The method using FFTs has some disadvantages. The first is that thecalculation amount is enormous, and the second is that peculiar noisesounds which cause uncomfortable feeling termed musical noises aregenerated.

On the other hand, in the method using band pass filters as shown inFIG. 12, the calculation amount can be suppressed to be small andmusical noises are not generated in principle. Further, there is anadvantage in that variation in the quality and quantity of the processcan be processed without making large changes.

As one reason behind this, since FFT can treat only the sample number of2², for example, the calculation amount is discrete and is not increasedby a small amount because there is a calculation resource. On the otherhand, with respect to band pass filters, since a single band pass filterhas a small unit of calculation amount, there are advantages in that thenumber of bands is easily increased and decreased and can be set indetail according to the calculation resource. Therefore, it isconsidered that the method using band pass filters is preferable.

However, in the method using band pass filters, there is a problem inthat sound quality is decreased compared with before the process isperformed.

When sounds are generally analyzed and combined using band pass filters,a method of analyzing the sound data of each of the bands divided byband pass filters, performing a process on the sound data of each of thebands in parallel, and finally combining all the sound data is used.

In a method of analyzing and combining sounds using band pass filters asshown in FIG. 12, sound quality is better than the case of FFTs.However, phase rotation is controlled and adjusted according to bandpass filters, addition/non-addition is controlled and adjusted accordingto bands, or increase/decrease in a level is controlled and adjusted.Therefore, when addition is performed for the respective bands, phasesmay not be matched compared to the original sound source, so that soundquality deterioration felt as noises is undeniable and becomes aproblem.

It is desirable to provide a signal processing method (noise suppressionmethod based on beam forming) for reducing noises while maintainingsound quality with respect to sound signals obtained from a plurality ofmicrophones, thereby improving calculation processing efficiency.

A sound signal processing apparatus according to an embodiment of thedisclosure includes a sound source direction determination unit thatdetermines sound source directions with respect to sound signals of aplurality of channels, which can be obtained by, for example, aplurality of microphone inputs or line inputs, for respective first ton-th bands; and a filter processing unit that includes first to n-thfilters which are connected in series and configured to boost orattenuate the sound signals with respect to the first to n-th bands. Therespective first to n-th filters perform boosting or attenuation basedon the sound source directions of the first to n-th bands which aredetermined by the sound source direction determination unit.

Further, the sound source direction determination unit may include firstto n-th sound source directional angle analysis units corresponding tothe first to n-th bands. Each of the first to n-th sound sourcedirectional angle analysis units may have one-to-one correspondence witheach of the first to n-th filters and regards the corresponding filtersas control targets for a boosting or attenuating process. Each of thefirst to n-th sound source directional angle analysis units may allowthe filter to be controlled to perform the boosting process when a soundsource direction of a corresponding band is determined as a directionincluded in a predetermined angle range, and allow the filter to becontrolled to perform the attenuating process when a sound sourcedirection angle of the corresponding band is not determined as adirection included in the predetermined angle range.

Further, each of the first to n-th sound source directional angleanalysis units may allow the filter to be controlled to perform theattenuating process when the sound source direction is determined be ina dispersion state.

Further, each of the first to n-th sound source directional angleanalysis units may determine the sound source direction with respect tothe corresponding band based on energy subtraction of the sound signalsof the respective channels.

Further, each of the first to n-th filters of the filter processingunit, which are connected in series, may receive a sound signal withwhich the sound signals of the plurality of channels are combined.

Further, each of the first to n-th filters, which are connected inseries, of the filter processing unit may receive a sound signal of oneof the plurality of channels.

A sound signal processing method according to another embodiment of thedisclosure may include determining sound source directions with respectto sound signals of a plurality of channels for respective first to n-thbands; and inputting sound signals to first to n-th filters which areconnected in series and configured to boost or attenuate the soundsignals with respect to the first to n-th bands, and performing boostingor attenuation by the respective first to n-th filters based on thesound source directions of the first to n-th bands, which are determinedin the determining of the sound source directions.

The above-described disclosure is a signal processing method (a noisesuppression method based on beam forming) for reducing noises whilemaintaining sound qualities of the apparatus which use two or moremicrophones, thereby improving calculation processing efficiency.

In order to remedy the deterioration in sound qualities accompanied by anoise reduction method, sound signals obtained by a single or two ormore separated microphones are divided for a respective plurality ofbands, and analysis (sound source direction determination) is performedto determine noises for the respective bands.

Thereafter, one or a plurality of additional values of the input soundsignals are processed based on the analysis results of the sound sourcedirection using a group of filters arranged in series on a time axis onwhich the mismatch of phases does not occur, thereby reducing noises.

The group of filters connected in series includes a plurality of bandboosting or attenuation filters capable of controlling gains, and thefilters are controlled based on the analysis results.

According to the disclosure, the sound signal process of reducing noiseswhile maintaining sound qualities with respect to sound signals obtainedfrom a plurality of microphones, thereby improving calculationprocessing efficiency, can be realized.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram of a noise suppression device according to anembodiment of the disclosure;

FIG. 2 is an explanatory view of MPF characteristics according to theembodiment;

FIG. 3 is an explanatory view of sample plots at the time of soundsource direction determination according to the embodiment;

FIGS. 4A to 4C are explanatory views of the sound source directiondetermination according to the embodiment;

FIG. 5 is an explanatory view of MPF control based on the sound sourcedirection determination according to the embodiment;

FIG. 6 is a flowchart of a process performed by a sound sourcedirectional angle analysis unit according to the embodiment;

FIG. 7 is an explanatory view of an example applied to an NC headphoneaccording to the embodiment;

FIG. 8 is a block diagram of the NC headphone according to theembodiment;

FIG. 9 is a block diagram of a noise suppression device according toanother embodiment;

FIG. 10 is an explanatory view of a conversation state in noisyconditions;

FIGS. 11A and 11B are explanatory views of a beam forming process; and

FIG. 12 is a block diagram of a noise suppression device according tothe related art.

DETAILED DESCRIPTION OF EMBODIMENTS

Hereinafter, embodiments of the disclosure will be described in thefollowing order.

1. Noise suppression device according to Embodiment

2. Example applied to NC headphone

3. Examples applied to Various Kinds of Apparatus and Modified Example

1. Noise Suppression Device According to Embodiment

FIG. 1 shows a noise suppression device 1 as an embodiment of a soundsignal processing apparatus of the disclosure.

The noise suppression device 1 obtains sound signals which arepreferable to a conversation in a noise environment in such a way thatsound signals absorbed by the left and right microphones 2L and 2R areinput, and sounds from, for example, the front (or the back), areboosted, and sounds from the other directions are attenuated.

In FIG. 1, a sound signal SmL obtained by the microphone 2L is amplifiedby the microphone amplifier 3L, and is converted to digital data by anAnalog-to-Digital (A/D) converter 4L. Thereafter, the sound signal SmLwhich is converted to the digital data is input to the noise suppressiondevice 1.

Further, a sound signal SmR obtained by the microphone 2R is amplifiedby the microphone amplifier 3R, and is converted to digital data by anA/D converter 4R. Thereafter, the sound signal SmR which is converted tothe digital data is input to the noise suppression device 1.

The noise suppression device 1 is configured to include a sound sourcedirection determination unit 1A and a filter processing unit 1B.

The sound source direction determination unit 1A determines the soundsource directions of the sound signals SmL and SmR of L and R channelsfor respective first to third bands in this example.

The filter processing unit 1B includes first to third filters (MPFs 58,59, and 60 which will be described later) that are configured to boostor attenuate the sound signals for the respective first to third bandsand are connected to each other in series.

The sound source direction determination unit 1A includes band passfilters 51L, 52L, 53L, 51R, 52R and 53R, and sound source directionalangle analysis units 54, 55, and 56.

The central pass frequencies of the respective band pass filters 51L,52L, and 53L are set to fc1, fc2, and fc3. For the explanation, therespective pass bands are represented as BD1, BD2, and BD3.

Further, the central pass frequencies of the respective band passfilters 51R, 52R, and 53R are set to central pass frequencies fc1, fc2,and fc3. The respective pass bands are represented as BD1, BD2, and BD3in the same manner.

The sound signal SmL of the left channel is input to the band passfilters 51L, 52L, and 53L, and the sound signal components of therespective bands BD1, BD2, and BD3 are extracted.

Further, the sound signal SmR of the right channel is input to the bandpass filters 51R, 52R, and 53R, and the sound signal components of therespective bands BD1, BD2, and BD3 are extracted.

The sound signal components of the band BD1 of each of the left andright channels, which are the outputs of the band pass filters 51L and51R, are provided to the sound source directional angle analysis unit54.

The sound signal components of the band BD2 of each of the left andright channels, which are the outputs of the band pass filters 52L and52R, are provided to the sound source directional angle analysis unit55.

The sound signal components of the band BD3 of each of the left andright channels, which are the outputs of the band pass filters 53L and53R, are provided to the sound source directional angle analysis unit56.

The sound source directional angle analysis unit 54 corresponds to theband BD1, and determines the sound source direction of a dominant soundfrom among the supplied sound signal components of the band BD1.

The sound source directional angle analysis unit 55 corresponds to theband BD2, and determines the sound source direction of a dominant soundfrom among the supplied sound signal components of the band BD2.

The sound source directional angle analysis unit 56 corresponds to theband BD3, and determines the sound source direction of a dominant soundfrom among the supplied sound signal components of the band BD3.

Although a method of determining the sound source directions by thesound source directional angle analysis units 54, 55, and 56 will bedescribed later, each of the sound source directional angle analysisunits 54, 55, and 56 determines the sound source direction based on theenergy subtraction of the sound signals of the respective channels withrespect to the corresponding bands.

Thereafter, the sound source directional angle analysis units 54, 55,and 56 control Mid Presence Filters (MPFs) 58, 59, and 60, which areprovided to has one-to-one correspondence with the sound sourcedirectional angle analysis units 54, 55, and 56, using control signalsSG1, SG2, and SG3 according to the determined directions. As can beunderstood from the drawing, an MPF 58 serves as a control target of thesound source directional angle analysis unit 54, an MPF 59 serves as acontrol target of the sound source directional angle analysis unit 55,and an MPF 60 serves as a control target of the sound source directionalangle analysis unit 56, respectively.

The filter processing unit 1B includes an adder 57, and MPFs 58, 59, and60. The MPFs 58, 59, and 60 serve as a series-connected filter group.

An adder 57 adds the sound signals SmL and SmR of the left and rightchannels. A sound signal (LR addition signal), in which the soundsignals of the left and right channels are combined by the adder 57, issupplied to the MPF 58.

The MPFs 58, 59, and 60 boost or attenuate the corresponding bands,respectively. Here, the reason why three MPFs are provided is that theband pass filters 51L, 52L, 53L, 51R, 52R, and 53R of the sound sourcedirection determination unit 1A divide each of the sound signals SmL andSmR into three bands.

The central pass frequencies of the respective MPFs 58, 59, and 60 areset to fc1, fc2, and fc3. Further, each of the MPFs 58, 59, and 60 hascharacteristics as shown in FIG. 2, and performs amplification andreduction of gain with respect to a specific target band (a bandcentering on a frequency fc). The boosting and attenuation of a targetband attributable to such variable gain adjustment in the MPFs 58, 59,and 60 are controlled by the sound source directional angle analysisunits 54, 55, and 56, as described above.

That is, although the MPF 58 boosts and attenuates the band BD1 whichcenters on the frequency fc1, the MPF 58 corresponds to the band passfilters 51L and 51R and the sound source directional angle analysis unit54.

Further, although the MPF 59 boosts and attenuates the band BD2 whichcenters on the frequency fc2, the MPF 59 corresponds to the band passfilters 52L and 52R and the sound source directional angle analysis unit55.

Further, although the MPF 60 boosts and attenuates the band BD3 whichcenters on the frequency fc3, the MPF 60 corresponds to the band passfilters 53L and 53R and the sound source directional angle analysis unit56.

Thereafter, when the noise suppression device 1 sets a front (back)direction as a target direction, a band in which a sound sourcedirection is determined as the front (back) direction is boosted, and aband in which the sound source direction is determined as anotherdirection is attenuated. A boosting/attenuation level is based on thedetermination of a directional angle.

In the respective MPFs 58, 59, and 60, the sound signal (LR additionsignal) is boosted or attenuated based on the control by the soundsource directional angle analysis units 54, 55, and 56. Thereafter, theoutput of the MPF 60 is the output signal Sout of the noise suppressiondevice 1.

The determination process of the sound source directional angle analysisunits 54, 55, and 56 and the control with respect to the MPFs 58, 59, 60will be described.

FIG. 3 shows the plots of sample values at the time of the sound sourcedirection/angle determination performed by the sound source directionalangle analysis units 54, 55, and 56.

Although the components of the bands BD1, BD2, and BD3 of the soundsignals SmL and SmR are input to the respective sound source directionalangle analysis units 54, 55, and 56, the sound source directional angleanalysis units 54, 55, and 56 plot the amplitude values of therespective L and R channels.

The plot locations on the LR plane of FIG. 3 represent the energysubtraction of the sound signals SmL and SmR of the respective L and Rchannels.

First, the absolute values of the amplitude values of the L/R channelsof a target band are plotted on the LR plane of FIG. 3, and this processis repeated during a specific time period.

For example, as an input value at a certain time point t0, if it isassumed that the absolute value of the amplitude of the L channel is setto A1 and the absolute value of the amplitude of the R channel is set toA2, the input value is plotted as a sample SPt0 represented as the blackcircle. This process is sequentially performed on each of the timepoints t1, t2, . . . , and samples SPt1, SPt2, . . . are plotted asshown in the drawing.

If a plurality of samples SPs are plotted during a certain unit time(for example, determined as about 0.5 to 5 seconds), a straight line LLstarting from an original point is obtained using a least-squaresmethod. That is, a straight line in which the sum of the squares of thedistance from all the samples SPs becomes minimum, and the straight lineis set to the straight line LL.

The angle θ of the straight line LL is regarded as the angle of thesound source direction.

With respect to the sound signal of a certain band, when an angle θ(straight line LL) comes around the center of the LR plane (around 45°in the drawing), it can be considered that the difference in theamplitude values in the corresponding band is small and it can beconsidered that the sound source is equidistant from the right and left.That is, the front direction can be estimated as the sound sourcedirection.

On the other hand, when the angle θ (straight line LL) is inclined tothe longitudinal axis or inclined to the lateral axis of the LR plane,it can be considered that the difference in the right and left amplitudevalues of the sound of the band is large and it can be considered thatthe sound from the right direction side or the left direction side.

Here, for example, as shown as oblique line sections in FIGS. 4A to 4C,the area of the angle θ in which the straight line LL exists around 45°is regarded as a center area. The center area is an area in which thesound source direction is regarded as the front (or back). On the otherhand, a right area and a left area in the drawing correspond to areas inwhich the sound source directions are regarded as the right side and theleft side, respectively. In FIGS. 4A to 4C, black circles are the plotpoints of the sample SPs.

For example, when the state described with reference to FIG. 10 isconsidered, the sound source direction of a conversational sound fromanother person can be considered as a front direction. In this case, thesound signal component of bands in which the sound source direction isthe front direction can be estimated as, for example, the conversationalsound, that is, the sound which a user wants to hear. On the other hand,the sound signal component of bands in which the sound source directionis another direction can be estimated as a noise sound, that is, thesound signal component desired to be reduced.

In this case, when the angle θ exists within the range of the centerarea on the LR plane as shown in FIG. 4A, the band is determined as aconversational sound (voice sound).

Further, as shown in FIG. 4B, when the angle θ exists in the area otherthan the center area, that is, a right area (or a left area), on the LRplane, the probability of the sound being from the front is low, and thesound of the band is determined as a noise sound.

Meanwhile, there is a case in which the sound should be determined as anoise even when the angle θ exists in the center area. When the samplepoints are broadly dispersed on the LR plane as shown in FIG. 4C, thestraight line LL according to the least-squares method becomes a slopearound 45° and the angle θ is included in the center area.

The case where the dispersion degree is high as described above is thecase where noise sounds arrive widely from many or all directionsattributable to surrounding reflected sounds. For example, the casecorresponds to a case where sounds, including reflected sounds as in thecabin of an airplane, are heard from all the directions.

Here, when the dispersion degree is equal to or higher than apredetermined degree, the sound of the band is determined as a noiseeven when the angle θ is included in the center area.

As a specific example, if the sum of squares of a distance when thestraight line LL is obtained using a least squares method is equal to orgreater than a specific threshold, it can be determined that thedispersion degree is large. The reason for this is that the sum ofsquares of the distances from the respective samples SPs to the straightline LL becomes small when the plotted dots of samples are concentratedin the center area, and the sum of squares becomes large in the caseshown in FIG. 4C.

FIG. 5 shows an example of the control of each of the sound sourcedirectional angle analysis units 54, 55, and 56.

Here, when the sound source directional angle analysis unit 54 performsthe above-described analysis on the band BD1, the angle θ of thestraight line LL is included in the center area. Although the soundsource directional angle analysis unit 54 controls the MPF 58 using thecontrol signal SG1 as described above, in this case, the sound of theband BD1 is determined as a target sound in this case, so that the bandBD1 which centers on the frequency fc1 is boosted by the MPF 58 as shownin the drawing.

Further, when the sound source directional angle analysis unit 55performs the above-described analysis on the band BD2, the angle θ ofthe straight line LL is in an area other than the center area. Althoughthe sound source directional angle analysis unit 55 controls the MPF 59using the control signal SG2, the sound of the band BD2 is determined asa noise in this case, so that the band BD2 which centers on thefrequency fc2 is attenuated by the MPF 59 as shown in the drawing.

Further, when the sound source directional angle analysis unit 56performs the above-described analysis on the band BD3, the angle θ ofthe straight line LL is included in the center area. However, since thedispersion degree of the sample dots becomes equal to or greater than apredetermined degree, the sound of the band BD3 is determined as anoise. Although the sound source directional angle analysis unit 56controls the MPF 60 using the control signal SG3, the sound of the bandBD3 is determined as a noise in this case, so that the band BD3 whichcenters on the frequency fc3 is attenuated by the MPF 60 as shown in thedrawing.

The filter characteristics of the MPFs 58, 59, and 60 are variablycontrolled based on the above-described determination of the soundsource direction for the respective bands, so that the output signalSout processed by the MPFs 58, 59, and 60 becomes a sound signal inwhich a sound from the front is boosted and the other noise isattenuated.

The processes of the above-described sound source directional angleanalysis units 54, 55, and 56 are performed as shown in FIG. 6. Theprocess of the sound source directional angle analysis unit 54 will bedescribed.

First, the sound source directional angle analysis unit 54 plots theinput values of the sound signals SmL and SmR of the band BD1 to beinput on the above-described LR plane during a specific unit time insteps F101 and F102.

After plotting a plurality of sample dots during the unit time, thesound source directional angle analysis unit 54 proceeds to step F103,obtains a straight line LL using a least-squares method, and thenobtains the angle θ of the straight line LL.

In step F104, it is first determined whether the angle θ is included inthe center area or not. If the angle θ is not included in the centerarea, the sound source directional angle analysis unit 54 proceeds tostep F107, and then determines that the sound of the corresponding bandBD1 is a noise. Thereafter, the sound source directional angle analysisunit 54 allows the MPF 58 to perform an attenuation process on the bandBD1 using the control signal SG1.

Meanwhile, it can be considered that the amount of the attenuation inthis case becomes, for example, the amount of the attenuation accordingto the difference between the angle θ at this time and the central angle(for example, 45° of the center area.

On the other hand, if it is determined that the angle θ is included inthe center area in step F104, the sound source directional angleanalysis unit 54 proceeds to step F105 and determines whether thedispersion state is equal to or greater than a specific level. Asdescribed above, it may be determined whether the sum of squares of thedistance between each of the samples and the straight line LL is equalto or greater than a specific threshold.

When the dispersion state is equal to or greater than the specificvalue, the sound source directional angle analysis unit 54 proceeds tostep F108, and determines that the sound of the corresponding band BD1is a noise. Thereafter, the sound source directional angle analysis unit54 allows the MPF 58 to perform the attenuation process on the band BD1using the control signal SG1.

Meanwhile, the amount of attenuation in this case can be considered as,for example, the amount of attenuation based on the value of the sum ofsquares of the distance.

When it is determined that the angle θ is included in the center areaand the dispersion state is not equal to or greater than the specificvalue, the sound source directional angle analysis unit 54 proceeds tostep F106, and determines that the sound of the corresponding band BD1is a target sound. Thereafter, the sound source directional angleanalysis unit 54 allows the MPF 58 to perform a boosting process on theband BD1 using the control signal SG1.

Meanwhile, the amount of boosting in this case can be considered as, forexample, the amount of boosting based on the difference between theangle θ at this time and the central angle (for example, 45°) of thecenter area and based on the dispersion degree.

That is, the amount of boosting is greater as the angle θ becomes closerto 45°, and the amount of boosting is greater as the dispersion degreebecomes smaller.

When any control is performed in steps F106, F107, and F108, the soundsource directional angle analysis unit 54 clears the plotted samples instep F109, and then returns to step F101 and performs plotting duringthe unit time again. Thereafter, the same process is repeated.

The sound source directional angle analysis unit 54 repeatedly andcontinuously performs the above-described process. The sound sourcedirectional angle analysis unit 55 and 56 perform the same process.

Therefore, for each unit time, the sound source direction of each bandis determined and the MPFs 58, 59, and 60 control filter characteristicsbased on the determination.

As understood from the above description, the noise suppression device 1in the present example divides the input sound signals SmL and SmR intobands BD1, BD2, and BD3 using the band pass filters 51L, 52L, 53L, 51R,52R, and 53R. Thereafter, analysis is performed to determine whether thesound signals are noises or not for the respective bands BD1, BD2, andBD3 using the sound source directional angle analysis units 54, 55, and56. On the other hand, the sound signals SmL and SmR are added andsupplied to the MPFs 58, 59, and 60 which are connected in series. Thefilter characteristic of each of the MPFs 58, 59, and 60 is variablycontrolled based on the determination results of the sound sourcedirectional angle analysis units 54, 55, and 56.

In this case, the control of a sound stream is a serial filter processand has the same system as a so-called equalizer. Therefore, thedeterioration in sound quality due to phase mismatch generated in theabove-described configuration of FIG. 12 does not occur in principle.Therefore, the output signal Sout which has no deterioration in soundquality can be obtained.

Further, since an FFT process is not used, the amount of calculation canbe suppressed to be low.

Further, the band pass filters can be scalably designed according to theamount of resources to be used. It is difficult to be realized in aprocess which uses FFTs.

In addition, the whole system can be mounted with little delay, and canbe applied to a field, in particular, to voice communication or thelike, which requests an extremely rapid response.

2. Example Applied to NC Headphone

An example in which the noise suppression device 1 according to theabove-described embodiment is applied to a noise cancellation headphone10 will be described.

FIG. 7 schematically shows the noise cancellation headphone (NCheadphone) 10 used while being connected to a music reproduction devicesuch as a portable media player 20 or the like.

The media player 20 reproduces data, such as music or the like, recordedin an internal recording medium, and outputs sound signals of two L andR channels to the connected NC headphone 10.

The NC headphone 10 includes a headphone unit 11 and a noisecancellation unit 14.

The headphone unit 11 includes the speakers 13L and 13R of the L and Rchannels in respective speaker housings corresponding to both left andright ears of a user.

In the case of this example, a noise cancellation process according to aso-called feedforward method is performed, and microphones 2L and 2R areprovided to collect the external sounds of the respective left and rightspeaker housings.

Meanwhile, the headphone unit 11 may not be the type having the speakerhousing as shown in the drawing but may be the types such as an earphonetype or an ear pad type. In the case of the present example, themicrophones 2L and 2R may be provided in any case.

The noise cancellation unit 14 is connected to the headphone unit 11 towhich the microphones 2L and 2R are provided as described above. Amonitor switch 43 is provided in the noise cancellation unit 14, so thata user can perform the on/off operation of a monitor mode.

Meanwhile, the monitor mode referred to here is a mode that enables aconversational sound or the like to be successfully heard while theoutput of music or the like which is being reproduced in the mediaplayer 20 is stopped and the noise cancellation function is turned on.

The noise cancellation unit 14 mixes a sound signal, such asreproduction music or the like, supplied from the media player 20 with anoise reduction sound signal, so that the sound signal from which theexternal noises are reduced is output from the speakers 13L and 13R.

Briefly speaking, noise reduction is performed as follows.

The microphones 2L and 2R mounted on the speaker housing collect anexternal noise which reaches the ears of a user through the speakerhousing. The noise cancellation unit 14 generates a noise reductionsound signal which has an acoustically reserved phase with respect tothat of the external noise based on the sound signal of the externalnoise collected by the microphones 2L and 2R. Thereafter, the noisecancellation unit 14 combines the generated noise reduction sound signalwith the sound signal, such as reproduction music or the like, and thensupplies the resulting signal to the speakers 13L and 13R.

Therefore, since the reserved phase component of the external noise isincluded in the sound output from the speakers 13L and 13R, the reservedphase component and the external noise actually leaked through thespeaker housing spatially are balanced out, so that the external noisecomponent is reduced and the output sound of the original reproductionmusic reaches the sense of hearing of the user.

The internal configuration of the noise cancellation unit 14 is shown inFIG. 8.

The noise cancellation unit 14 includes microphone amplifiers 3L and 3R,A/D converters 4L and 4R, a main processing unit 33 based on a DSP or aCPU, a memory unit 40, power amplifiers 42L and 42R, A/D converters 41Land 41R, and a monitor switch 43.

In the main processing unit 33, a noise cancellation unit 34, a gainunit 35, adders 36L and 36R, a noise suppression device 1, a controlunit 38, an equalizer 39, and switches SW1 and SW2 are provided.

First, a sound signal, such as reproduction music or the like, from themedia player 20 is processed as follows.

The reproduction sound signals SA-L and SA-R of the L and R channels aresupplied from the media player 20 as a so-called headphone output.

The reproduction sound signals SA-L and SA-R are converted into digitalsignals by the A/D converters 41L and 41R. Thereafter, the equalizer 39performs sound quality correction such as amplitude-frequencycharacteristic correction, phase-frequency characteristic correction, orboth corrections.

The correction process of the equalizer 39 is performed based on acontrol signal from the control unit 38. For example, the indication ofa frequency characteristic or the like is performed using the controlsignal.

The reproduction sound signals SA-L and SA-R in which the soundqualities thereof are corrected by the equalizer 39 are respectivelyprovided to the adders 36L and 36R through the switches SW1 and SW2connected to a Te terminal. Thereafter, the reproduction sound signalsSA-L and SA-R are added to noise reduction sound signals by the adders36L and 36R, and then the resulting signals are supplied to the poweramplifiers 42L and 42R.

The power amplifiers 42L and 42R may be configured with digitalamplifiers, and may be configured with a D/A converter and an analogamplifier.

Further, the output from the power amplifiers 42L and 42R serve asdriving signals corresponding to the speakers 13L and 13R, and soundsare output from the speakers 13L and 13R based on the reproduction soundsignals SA-L and SA-R.

On the other hand, a process for the above-described noise cancellationis performed as follows.

The sound signals SmL and SmR collected by the microphones 2L and 2R areamplified by the microphone amplifiers 3L and 3R of the noisecancellation unit 14, and then converted into digital signals by the A/Dconverters 4L and 4R.

The sound signals SmL and SmR which are converted into the digitalsignals and output from the A/D converters 4L and 4R are supplied to thenoise cancellation unit 34. The noise cancellation unit 34 serves as adigital filter that generates the above-described noise reduction soundsignals using the feedforward method. The noise cancellation unit 34performs a filtering process on the respective sound signals SmL and SmRusing a filter coefficient as instructed by the control signal from thecontrol unit 38, and then generates the noise reduction sound signals ofthe L and R channels.

The generated noise reduction sound signals of the L and R channels aresupplied to the gain unit 35. The gain unit 35 assigns gainscorresponding to the noise reduction sound signals of the L and Rchannels using the gain coefficient as instructed by the control signalfrom the control unit 38.

Thereafter, the noise reduction sound signals of the L and R channelsfrom the gain unit 35 are added to the respective reproduction soundsignals SA-L and SA-R which are supplied to the adders 36L and 36R asdescribed above.

The reproduction sounds are output from the speakers 13L and 13R basedon the reproduction sound signals SA-L and SA-R to which the noisereduction sound signals are added, so that the above-described noisereduction function is exerted.

The control unit 38 controls the whole noise cancellation unit. Forexample, the control unit 38 controls the equalizer 39, the noisecancellation unit 34, and the gain unit 35 using the control signal asdescribed above. Further, the control unit 38 can transmit the controlsignal to the media player 20. Further, the control unit 38 controls theswitching of the switches SW1 and SW2.

The memory unit 40 stores information which is referred to when thecontrol unit 38 performs a control process. For example, the memory unit40 stores information about the filter coefficients of the noisecancellation unit 34 and the equalizer 39 or the like.

The noise cancellation unit 14 of the present example further includesthe noise suppression device 1 which has the configuration as describedin FIG. 1.

The sound signals SmL and SmR which are converted into digital signalsand output from the A/D converters 4L and 4R are supplied to the noisesuppression device 1. The noise suppression device 1 performs theconfiguration and operation described with reference to FIGS. 1 to 6 onthe input sound signals SmL and SmR.

Therefore, the output signal Sout, in which the sound from the frontdirection is boosted as a target sound, such as a conversational soundor the like, and sounds from the other directions are attenuated, isobtained from the noise suppression device 1. The output signal Sout issupplied to the Tn terminals of the switches SW1 and SW2.

Particularly, in the present example, when the control unit 38 detectsthat a user turns on a monitor mode using the monitor switch 43, thecontrol unit 38 performs control as follows.

If the monitor mode is turned on, the control unit 38 switches theswitches SW1 and SW2 to the Tn terminals. Meanwhile, when the monitormode is turned off, the control unit 38 connects the switches SW1 andSW2 to the Te terminals, and reproduction music is output from thespeakers 13L and 13R.

Further, the control unit 38 instructs the media player 20 to stop thereproduction music. Therefore, the media player 20 stops thereproduction music.

When the control unit 38 performs the above-described control, theoutput signal Sout of the noise suppression device 1 is supplied to theadder 36L and 36R.

Therefore, the noise reduction sound signals from the gain unit 35 andthe output signal Sout from the noise suppression device 1 are added bythe adders 36L and 36R, and then the resulting signals are supplied tothe power amplifiers 42L and 42R. Thereafter, the resulting signals areoutput from the speakers 13L and 13R as sounds.

These sounds become speaker output sounds in which, for example, aconversational sound from the front direction is clearly audible whilesurrounding noises are reduced as in the monitor mode.

When the noise suppression device 1 according to the present embodimentis mounted on the NC headphone 10 as described above, the speaker outputin which a conversational sound is clearly audible can be realized as amonitor mode operation.

That is, when the NC headphone 10 is used, not only noise but also thesound of a person is reduced. However, with the above-describedconfiguration, surrounding noises can be reduced while the sound of aperson who is at the front which is equidistance from the microphones 2Land 2R is not reduced. Therefore, conversation can be performed morepleasantly while wearing the NC headphone 10.

On that basis, in the noise suppression device 1, control on a soundstream is performed by a series filtering process as described above,and the deterioration in sound qualities due to phase mismatching doesnot occur, so that sound can be output without deterioration in soundquality.

Further, with a low calculation amount and a process using lowresources, it is suitable for mounting on a small device such as thenoise cancellation unit 14 or the like.

In addition, the whole system can be mounted with low delay.

With respect to the monitor mode function of the NC headphone 10, anactual direct sound and the sound obtained after the process of thenoise suppression device 1 is performed are spatially superimposed andthen reach the ears of a user. Therefore, if the process delay is large,the sounds are heard as unpleasant echo. However, since the noisesuppression device 1 can perform a process with low delay, such anunpleasant echo can be avoided.

3. Examples Applied to Various Kinds of Apparatus and Modified Example

The noise suppression device 1 according to the present embodiment canbe applied to further various kinds of apparatus.

For example, it can be considered that the noise suppression device 1 isused for a transmission noise reduction function of a mobile phone.

By mounting the noise suppression device 1 on a headset for a mobilephone, the sound can be transmitted to a counter side while the sound,emitted from the mouth of a user which is at equidistance from themicrophones, is not reduced and surrounding noises are reduced.

Of course, the voice communication performed using a Personal Computer(PC) or a television receiver is the same.

Further, an application to a sound recognition front-end may beconsidered.

Nowadays, a sound recognition function-attached “automatic translation”or the like, which is used in a mobile phone or a small-sized PersonalComputer (PC), has reached a workable level which can be ordinarilyused, and, hereinafter, it can be considered that such a function can beused outside. On the other hand, when a sound is input outdoors, noiseswhich deteriorate the accuracy of sound recognition may be input in manycases.

Therefore, for example, if a front-end process is performed using thenoise suppression device 1 according to the present embodiment in such away that microphones are attached, for example, on both ends of portableapparatus, an automatic translation system becomes a system with whichthe user is satisfied.

Further, application as a system for extracting a vocal sound or thelike can be considered.

Although an application as the microphone input is described in theabove-described embodiment, application to line input or a music filecan be considered.

For example, since a vocal sound, a drum sound, or the like is made tobe stereotaxically centered in general music, if the noise suppressiondevice 1 according to the present embodiment is applied thereto, thevocal sound and the drum sound can be separated. Thereafter, of course,if bands are divided, the separation of the vocal sound and the drumsound can be performed.

A modified example of the embodiment can be variously considered.

FIG. 9 shows a modified example of the configuration of the noisesuppression device 1. This is an example in which a group of twoindependent systems of series filters is provided in an L channel and anR channel.

That is, the sound signal SmL of the L channel is input to series filtersystems of MPFs 58L, 59L, and 60L. The sound signal SmR of the R channelis input to series filter systems of MPFs 58R, 59R, and 60R.

The filter characteristics of the MPFs 58L and 58R are variablycontrolled using a control signal SG1 based on the determination of asound source directional angle analysis unit 54.

The filter characteristics of the MPFs 59L and 59R are variablycontrolled using a control signal SG2 based on the determination of asound source directional angle analysis unit 55.

The filter characteristics of the MPFs 60L and 60R are variablycontrolled using a control signal SG3 based on the determination of asound source directional angle analysis unit 56.

That is, the modified example is a configuration example in which theoperation thereof is the same as in the configuration of FIG. 1 butoutput signals SoutL and SoutR of the two L and R channels are output assignals obtained after a process is performed.

The noise suppression device 1 may be applied to various types ofapparatus with such a configuration.

Further, although not shown in the drawing, sound source directiondetermination to be performed on microphone input sounds of three ormore channels can be considered.

In such a case, sound signals on which a series filter processes isperformed may be combined into a single channel or two channels as shownin FIG. 8. Further, the output signals Souts of three or more channelsmay be obtained in such a way that the series filter process isperformed on each of the microphone input sound signals of three or morechannels, independently.

Further, it may be considered that the sound signal of a single channelis supplied from among the input sound signals of a plurality ofchannels when a single series filter system is provided. For example, itmay be a configuration in which only the sound signal SmL is supplied tothe filter group of the MPFs 58, 59, and 60 and the output signal Soutis obtained when the single series filter system (MPFs 58, 59, and 60)is provided as shown in FIG. 1.

Further, it is proper that the number of bands divided by the band passfilters and the bandwidth of a single band are set according to theapparatus to be mounted, a target sound, a usage form, or the like. Thenumber of MPFs connected in series is basically set according to thenumber of bands divided by the band pass filters.

Further, although the process of boosting the sound from the frontdirection or the back direction as a target sound is described in theembodiment, for example, a process of boosting a sound from the rightside as a target sound and reducing sounds from the other directions canbe performed. For example, when the angle θ corresponds to the rightarea as shown in FIG. 4B, a boosting process may be performed on the MPFcorresponding to the band of the target sound, and an attenuatingprocess may be performed on the MPFs corresponding to the bands in whichthe angle θ corresponds to the center area and the left area.

That is, the sound source direction of a target sound can be set usingany method.

Further, although the noise suppression device 1 performs the digitaldata process using the A/D converters 4L and 4R in the embodiment asshown in FIG. 1, the filtering process performed by the MPFs 58, 59, and60 or the band division performed by the band pass filters may beperformed using an analog signal process.

The present disclosure contains subject matter related to that disclosedin Japanese Priority Patent Application JP 2010-125502 filed in theJapan patent office on Jun. 1, 2010, the entire contents of which arehereby incorporated by reference.

It should be understood by those skilled in the art that variousmodifications, combinations, sub-combinations and alterations may occurdepending on design requirements and other factors insofar as they arewithin the scope of the appended claims or the equivalents thereof.

1. A sound signal processing apparatus comprising: a sound sourcedirection determination unit that determines sound source directionswith respect to sound signals of a plurality of channels for respectivefirst to n-th bands; and a filter processing unit that includes first ton-th filters which are connected in series and configured to boost orattenuate the sound signals with respect to the first to n-th bands,wherein the respective first to n-th filters perform boosting orattenuation based on the sound source directions of the first to n-thbands which are determined by the sound source direction determinationunit.
 2. The sound signal processing apparatus according to claim 1,wherein the sound source direction determination unit includes first ton-th sound source directional angle analysis units corresponding to thefirst to n-th bands, wherein each of the first to n-th sound sourcedirectional angle analysis units has one-to-one correspondence with eachof the first to n-th filters, and regards the corresponding filters ascontrol targets for a boosting or attenuating process, and wherein eachof the first to n-th sound source directional angle analysis unitsallows the filter to be controlled to perform the boosting process whena sound source direction of a corresponding band is determined as adirection included in a predetermined angle range, and allows the filterto be controlled to perform the attenuating process when a sound sourcedirection angle of the corresponding band is not determined as adirection included in the predetermined angle range.
 3. The sound signalprocessing apparatus according to claim 2, wherein each of the first ton-th sound source directional angle analysis units allows the filter tobe controlled to perform the attenuating process when the sound sourcedirection is determined to be in a dispersion state.
 4. The sound signalprocessing apparatus according to claim 3, wherein each of the first ton-th sound source directional angle analysis units determines the soundsource direction with respect to the corresponding band based on energysubtraction of the sound signals of the respective channels.
 5. Thesound signal processing apparatus according to claim 1, wherein each ofthe first to n-th filters of the filter processing unit, which areconnected in series, receives a sound signal with which the soundsignals of the plurality of channels are combined.
 6. The sound signalprocessing apparatus according to claim 1, wherein each of the first ton-th filters of the filter processing unit, which are connected inseries, receives a sound signal of one of the plurality of channels. 7.A sound signal processing method comprising: determining sound sourcedirections with respect to sound signals of a plurality of channels forrespective first to n-th bands; and inputting sound signals to first ton-th filters which are connected in series and configured to boost orattenuate the sound signals with respect to the first to n-th bands, andperforming boosting or attenuation by the respective first to n-thfilters based on the sound source directions of the first to n-th bands,which are determined in the determining of the sound source directions.